[Chromoting] Use ProcessStatsSender in host process

remoting/protocol/webrtc_connection_to_client.cc

 // Copyright 2015 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "remoting/protocol/webrtc_connection_to_client.h"
 
 #include <utility>
 
 #include "base/bind.h"
 #include "base/location.h"
 #include "jingle/glue/thread_wrapper.h"
 #include "net/base/io_buffer.h"
 #include "remoting/codec/video_encoder.h"
 #include "remoting/codec/webrtc_video_encoder_vpx.h"
 #include "remoting/protocol/audio_source.h"
 #include "remoting/protocol/audio_stream.h"
 #include "remoting/protocol/clipboard_stub.h"
 #include "remoting/protocol/host_control_dispatcher.h"
 #include "remoting/protocol/host_event_dispatcher.h"
 #include "remoting/protocol/host_stub.h"
 #include "remoting/protocol/input_stub.h"
 #include "remoting/protocol/message_pipe.h"
+#include "remoting/protocol/process_stats_dispatcher.h"
+#include "remoting/protocol/process_stats_stub.h"
 #include "remoting/protocol/transport_context.h"
 #include "remoting/protocol/webrtc_audio_stream.h"
 #include "remoting/protocol/webrtc_transport.h"
 #include "remoting/protocol/webrtc_video_stream.h"
 #include "third_party/webrtc/api/mediastreaminterface.h"
 #include "third_party/webrtc/api/peerconnectioninterface.h"
 #include "third_party/webrtc/api/test/fakeconstraints.h"
 
 namespace remoting {
 namespace protocol {
 
 // Currently the network thread is also used as worker thread for webrtc.
 //
 // TODO(sergeyu): Figure out if we would benefit from using a separate
 // thread as a worker thread.
 WebrtcConnectionToClient::WebrtcConnectionToClient(
     std::unique_ptr<protocol::Session> session,
     scoped_refptr<protocol::TransportContext> transport_context,
     scoped_refptr<base::SingleThreadTaskRunner> video_encode_task_runner,
     scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner)
     : transport_(
           new WebrtcTransport(jingle_glue::JingleThreadWrapper::current(),
                               transport_context,
                               this)),
       session_(std::move(session)),
       video_encode_task_runner_(video_encode_task_runner),
       audio_task_runner_(audio_task_runner),
       control_dispatcher_(new HostControlDispatcher()),
       event_dispatcher_(new HostEventDispatcher()),
+      stats_dispatcher_(new ProcessStatsDispatcher()),
       weak_factory_(this) {
   session_->SetEventHandler(this);
   session_->SetTransport(transport_.get());
 }
 
 WebrtcConnectionToClient::~WebrtcConnectionToClient() {}
 
 void WebrtcConnectionToClient::SetEventHandler(
     ConnectionToClient::EventHandler* event_handler) {
   DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 35 matching lines @@
   stream->Start(audio_task_runner_, std::move(audio_source), transport_.get());
   return std::move(stream);
 }
 
 // Return pointer to ClientStub.
 ClientStub* WebrtcConnectionToClient::client_stub() {
   DCHECK(thread_checker_.CalledOnValidThread());
   return control_dispatcher_.get();
 }
 
+// Return pointer to ProcessStatsStub.
+ProcessStatsStub* WebrtcConnectionToClient::stats_stub() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  return stats_dispatcher_.get();
+}
+
 void WebrtcConnectionToClient::set_clipboard_stub(
     protocol::ClipboardStub* clipboard_stub) {
   DCHECK(thread_checker_.CalledOnValidThread());
   control_dispatcher_->set_clipboard_stub(clipboard_stub);
 }
 
 void WebrtcConnectionToClient::set_host_stub(protocol::HostStub* host_stub) {
   DCHECK(thread_checker_.CalledOnValidThread());
   control_dispatcher_->set_host_stub(host_stub);
 }
@@ skipping 27 matching lines @@
       // being torn down.
       if (self)
         event_handler_->CreateMediaStreams();
       break;
     }
 
     case Session::CLOSED:
     case Session::FAILED:
       control_dispatcher_.reset();
       event_dispatcher_.reset();
+      stats_dispatcher_.reset();
       transport_->Close(state == Session::CLOSED ? OK : session_->error());
       transport_.reset();
       event_handler_->OnConnectionClosed(
           state == Session::CLOSED ? OK : session_->error());
       break;
   }
 }
 
 void WebrtcConnectionToClient::OnWebrtcTransportConnecting() {
   DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 48 matching lines @@
     ChannelDispatcherBase* channel_dispatcher) {
   DCHECK(thread_checker_.CalledOnValidThread());
 
   LOG(ERROR) << "Channel " << channel_dispatcher->channel_name()
              << " was closed unexpectedly.";
   Disconnect(INCOMPATIBLE_PROTOCOL);
 }
 
 }  // namespace protocol
 }  // namespace remoting